sip: media: rework codec negotiation

introduce `calls_sip_media_manager_get_capabilities ()` which takes
a GList of MediaCodecInfo's as input to generate a SDP message.
If using in an SDP answer we simply feed it a list of the common codecs
as gathered from the SDP offer.

plugins/sip/calls-sip-call.c

@@ -46,6 +46,7 @@ struct _CallsSipCall
   CallsSipMediaManager *manager;
   CallsSipMediaPipeline *pipeline;
   nua_handle_t *nh;
+  GList *codecs;
 };
 
 static void calls_sip_call_message_source_interface_init (CallsMessageSourceInterface *iface);
@@ -108,9 +109,10 @@ calls_sip_call_answer (CallsCall *call)
   /* TODO get free port by creating GSocket and passing that to the pipeline */
   calls_sip_call_setup_local_media_connection (self, local_port, local_port + 1);
 
-  local_sdp = calls_sip_media_manager_static_capabilities (self->manager,
-                                                           local_port,
-                                                           FALSE);
+  local_sdp = calls_sip_media_manager_get_capabilities (self->manager,
+                                                        local_port,
+                                                        FALSE,
+                                                        self->codecs);
 
   g_assert (local_sdp);
   g_debug ("Setting local SDP to string:\n%s", local_sdp);
@@ -210,6 +212,8 @@ calls_sip_call_finalize (GObject *object)
     calls_sip_media_pipeline_stop (self->pipeline);
     g_clear_object (&self->pipeline);
   }
+  g_clear_pointer (&self->codecs, g_list_free);
+
   G_OBJECT_CLASS (calls_sip_call_parent_class)->finalize (object);
 }
 
@@ -354,3 +358,14 @@ calls_sip_call_set_state (CallsSipCall   *self,
                          old_state);
 }
 
+
+void
+calls_sip_call_set_codecs (CallsSipCall *self,
+                           GList        *codecs)
+{
+  g_return_if_fail (CALLS_IS_SIP_CALL (self));
+  g_return_if_fail (codecs);
+
+  g_list_free (self->codecs);
+  self->codecs = codecs;
+}

plugins/sip/calls-sip-call.h

@@ -49,5 +49,7 @@ void                   calls_sip_call_activate_media                    (CallsSi
                                                                          gboolean      enabled);
 void                   calls_sip_call_set_state                         (CallsSipCall  *self,
                                                                          CallsCallState state);
+void                   calls_sip_call_set_codecs                        (CallsSipCall *self,
+                                                                         GList        *codecs);
 
 G_END_DECLS

plugins/sip/calls-sip-media-manager.c

@@ -85,18 +85,21 @@ calls_sip_media_manager_default ()
 }
 
 
-/* calls_sip_media_manager_static_capabilities:
+/* calls_sip_media_manager_get_capabilities:
  *
+ * @self: A #CallsSipMediaManager
  * @port: Should eventually come from the ICE stack
  * @use_srtp: Whether to use srtp (not really handled)
+ * @supported_codecs: A #GList of #MediaCodecInfo
  *
- * Returns: (full-control) string describing capabilities
+ * Returns: (transfer: full) string describing capabilities
  * to be used in the session description (SDP)
  */
 char *
-calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
-                                             guint                 port,
-                                             gboolean              use_srtp)
+calls_sip_media_manager_get_capabilities (CallsSipMediaManager *self,
+                                          guint                 port,
+                                          gboolean              use_srtp,
+                                          GList                *supported_codecs)
 {
   char *payload_type = use_srtp ? "SAVP" : "AVP";
   g_autoptr (GString) media_line = NULL;
@@ -108,7 +111,7 @@ calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
   media_line = g_string_new (NULL);
   attribute_lines = g_string_new (NULL);
 
-  if (self->supported_codecs == NULL) {
+  if (supported_codecs == NULL) {
     g_warning ("No supported codecs found. Can't build meaningful SDP message");
     g_string_append_printf (media_line, "m=audio 0 RTP/AVP 0");
     goto done;
@@ -118,7 +121,7 @@ calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
   g_string_append_printf (media_line,
                           "m=audio %d RTP/%s", port, payload_type);
 
-  for (node = self->supported_codecs; node != NULL; node = node->next) {
+  for (node = supported_codecs; node != NULL; node = node->next) {
     MediaCodecInfo *codec = node->data;
 
     g_string_append_printf (media_line, " %u", codec->payload_id);
@@ -141,8 +144,86 @@ calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
 }
 
 
+/* calls_sip_media_manager_static_capabilities:
+ *
+ * @self: A #CallsSipMediaManager
+ * @port: Should eventually come from the ICE stack
+ * @use_srtp: Whether to use srtp (not really handled)
+ *
+ * Returns: (transfer: full) string describing capabilities
+ * to be used in the session description (SDP)
+ */
+char *
+calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
+                                             guint                 port,
+                                             gboolean              use_srtp)
+{
+  g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
+
+  return calls_sip_media_manager_get_capabilities (self,
+                                                   port,
+                                                   use_srtp,
+                                                   self->supported_codecs);
+}
+
+
 MediaCodecInfo*
 get_best_codec (CallsSipMediaManager *self)
 {
   return media_codec_by_name ("PCMA");
 }
+
+
+/* calls_sip_media_manager_codec_candiates:
+ *
+ * @self: A #CallsSipMediaManager
+ *
+ * Returns: (transfer: none) A #GList of supported
+ * #MediaCodecInfo
+ */
+GList *
+calls_sip_media_manager_codec_candidates (CallsSipMediaManager *self)
+{
+  g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
+
+  return self->supported_codecs;
+}
+
+
+/* calls_sip_media_manager_get_codecs_from_sdp
+ *
+ * @self: A #CallsSipMediaManager
+ * @sdp: A #sdp_media_t media description
+ *
+ * Returns: (transfer: full) A #GList of codecs found in the
+ * SDP message
+ */
+GList *
+calls_sip_media_manager_get_codecs_from_sdp (CallsSipMediaManager *self,
+                                             sdp_media_t          *sdp_media)
+{
+  GList *codecs = NULL;
+  sdp_rtpmap_t *rtpmap = NULL;
+
+  g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
+  g_return_val_if_fail (sdp_media, NULL);
+
+  if (sdp_media->m_type != sdp_media_audio) {
+    g_warning ("Only the 'audio' media type is supported");
+    return NULL;
+  }
+
+  for (rtpmap = sdp_media->m_rtpmaps; rtpmap != NULL; rtpmap = rtpmap->rm_next) {
+    MediaCodecInfo *codec = media_codec_by_payload_id (rtpmap->rm_pt);
+    if (codec)
+      codecs = g_list_append (codecs, codec);
+  }
+
+  if (sdp_media->m_next != NULL)
+    g_warning ("Currently only a single media session is supported");
+
+  if (codecs == NULL)
+    g_warning ("Did not find any common codecs");
+
+  return codecs;
+}

plugins/sip/calls-sip-media-manager.h

@@ -27,6 +27,7 @@
 #include "gst-rfc3551.h"
 
 #include <sofia-sip/nua.h>
+#include <sofia-sip/sdp.h>
 #include <glib-object.h>
 
 #define CALLS_TYPE_SIP_MEDIA_MANAGER (calls_sip_media_manager_get_type ())
@@ -35,9 +36,16 @@ G_DECLARE_FINAL_TYPE (CallsSipMediaManager, calls_sip_media_manager, CALLS, SIP_
 
 
 CallsSipMediaManager* calls_sip_media_manager_default                       (void);
+gchar*                calls_sip_media_manager_get_capabilities              (CallsSipMediaManager *self,
+                                                                             guint port,
+                                                                             gboolean use_srtp,
+                                                                             GList *supported_codecs);
 gchar*                calls_sip_media_manager_static_capabilities           (CallsSipMediaManager *self,
                                                                              guint port,
                                                                              gboolean use_srtp);
 gboolean              calls_sip_media_manager_supports_media                (CallsSipMediaManager *self,
                                                                              const char *media_type);
 MediaCodecInfo*       get_best_codec                                        (CallsSipMediaManager *self);
+GList *               calls_sip_media_manager_codec_candidates              (CallsSipMediaManager *self);
+GList *               calls_sip_media_manager_get_codecs_from_sdp           (CallsSipMediaManager *self,
+                                                                             sdp_media_t          *sdp_media);

plugins/sip/calls-sip-origin.c

@@ -218,9 +218,14 @@ sip_i_state (int              status,
              tagi_t           tags[])
 {
   const sdp_session_t *r_sdp = NULL;
+  const sdp_session_t *l_sdp = NULL;
   gint call_state = nua_callstate_init;
   CallsCallState state;
   CallsSipCall *call;
+  int offer_sent = 0;
+  int offer_recv = 0;
+  int answer_sent = 0;
+  int answer_recv = 0;
 
   g_assert (CALLS_IS_SIP_ORIGIN (origin));
 
@@ -234,8 +239,13 @@ sip_i_state (int              status,
   }
 
   tl_gets (tags,
+           SOATAG_LOCAL_SDP_REF (l_sdp),
            SOATAG_REMOTE_SDP_REF (r_sdp),
            NUTAG_CALLSTATE_REF (call_state),
+           NUTAG_OFFER_SENT_REF (offer_sent),
+           NUTAG_OFFER_RECV_REF (offer_recv),
+           NUTAG_ANSWER_SENT_REF (answer_sent),
+           NUTAG_ANSWER_RECV_REF (answer_recv),
            TAG_END ());
 
   if (status == 503) {
@@ -247,6 +257,11 @@ sip_i_state (int              status,
    * also: rtcp port = rtp port + 1
    */
   if (r_sdp) {
+    GList *codecs =
+      calls_sip_media_manager_get_codecs_from_sdp (origin->media_manager,
+                                                   r_sdp->sdp_media);
+
+    calls_sip_call_set_codecs (call, codecs);
     calls_sip_call_setup_remote_media_connection (call,
                                                   r_sdp->sdp_connection->c_address,
                                                   r_sdp->sdp_media->m_port,
@@ -1161,7 +1176,7 @@ calls_sip_origin_go_online (CallsSipOrigin *self,
     nua_register (self->oper->register_handle,
                   SIPTAG_EXPIRES_STR ("180"),
                   NUTAG_SUPPORTED ("replaces, outbound, gruu"),
-                  NUTAG_OUTBOUND ("outbound natify gruuize"), // <- janus uses "no-validate"
+                  NUTAG_OUTBOUND ("outbound natify gruuize validate"), // <- janus uses "no-validate"
                   NUTAG_M_USERNAME (self->user),
                   NUTAG_M_DISPLAY ("SIP tester"),
                   NUTAG_M_PARAMS ("user=phone"),

plugins/sip/gst-rfc3551.c

@@ -55,6 +55,17 @@ media_codec_by_name (const char *name)
   return NULL;
 }
 
+MediaCodecInfo *
+media_codec_by_payload_id (guint payload_id)
+{
+  for (guint i = 0; i < G_N_ELEMENTS (gst_codecs); i++) {
+    if (payload_id == gst_codecs[i].payload_id)
+      return &gst_codecs[i];
+  }
+
+  return NULL;
+}
+
 gchar *
 media_codec_get_gst_capabilities (MediaCodecInfo *codec)
 {

plugins/sip/gst-rfc3551.h

@@ -44,5 +44,6 @@ typedef struct {
 
 
 MediaCodecInfo*  media_codec_by_name (const char *name);
+MediaCodecInfo*  media_codec_by_payload_id (uint payload_id);
 gchar*           media_codec_get_gst_capabilities (MediaCodecInfo *codec);
 GList*           media_codecs_get_candidates ();